Method for enhancing audio signals

ABSTRACT

A frequency band of an audio input signal is analyzed to determine if a transient is present. When transients are detected, modifications are made to the intensity levels corresponding to the frequency band for a brief time period.

CROSS-REFERENCES TO RELATED APPLICATIONS

This application claims priority of provisional U.S. Patent ApplicationSer. No. 60/746,625, filed May 5, 2006, titled “Method for EnhancingAudio Signals” the disclosure of which is incorporated by reference inits entirety.

FIELD OF THE INVENTION

The present invention relates to audio signal processing. Moreparticularly, the present invention relates to enhancing selectedportions of an audio signal in response to detected characteristics ofthe signal.

BACKGROUND OF THE INVENTION

Audio signals or streams typically may be rendered to a listener, suchas by using a speaker to provide an audible rendering of the audiosignal or stream. An audio signal or stream so rendered may have one ormore characteristics that may be perceived and, in some cases,identified and/or described by a discerning listener. For example, alistener may be able to detect how sharply or clearly transient audioevents, such as a drumstick hitting a drum, are rendered.

Different audio playback equipment presents different renderingcharacteristics. For example, “high end” audio equipment may renderaudio in a different manner than lower cost audio equipment. In theformer case, the audio signals may be rendered in a way that permitsaudio characteristics such as transient audio events to be perceived toa greater extent than from the use of less expensive audio equipment.

Different listeners may have different preferences and/or tastes withrespect to such identifiable perceptual characteristics. One listenermay prefer that transient audio events, such as drum hits, be enhancedor otherwise emphasized, whereas another might instead prefer that suchtransient events be suppressed to some extent or otherwisede-emphasized. In addition, an individual listener may prefer that suchtransients be enhanced for certain types of audio data (e.g., rockmusic), and suppressed or softened to a degree for other types (e.g.,classical music or non-music recordings).

It is therefore desirable to provide improved control over transientsand other characteristics in an audio stream.

SUMMARY OF THE INVENTION

The present invention provides a method for enhancing thecharacteristics of an audio signal. An input audio signal is filteredinto at least one frequency band and decomposed into a sequence offrames, each frame corresponding to a group of samples from thesegmented signal. A flux value is determined for each frame and asequence of flux values generated. In one embodiment, the flux valuecorresponds to the RMS value of the frame or portion of the frameexamined. The sequence of flux values is analyzed to determine a firstorder energy level variation exceeding a threshold. Where there exists arapid change in flux values, a corresponding portion of the signal isbriefly enhanced.

In accordance with one embodiment, a method of enhancing an audio signalis provided. A sequence of flux values is derived from the input signal.The method comprises monitoring the energy level variations in at leastone frequency band of the audio signal and applying a dynamic timevarying equalizer to the audio signal based on the monitored energylevel variations. In one embodiment, monitoring the energy levelvariations comprises decomposing the audio signal into a sequence offrames, each frame having at least one sampled value and assigning anenergy level for each frame in the sequence.

These and other features and advantages of the present invention aredescribed below with reference to the drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram illustrating a system and method fordynamically enhancing an audio signal in accordance with one embodimentof the present invention.

FIG. 2 is a block diagram illustrating a system for enhancing an audiosignal in accordance with one embodiment of the present invention.

FIG. 3 is a flow chart illustrating a method of enhancing an audiosignal in accordance with one embodiment of the present invention.

FIG. 4 is a diagram illustrating a transfer function for mapping fluxvalues to a control signal gain in accordance with one embodiment of thepresent invention.

DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS

Reference will now be made in detail to preferred embodiments of theinvention. Examples of the preferred embodiments are illustrated in theaccompanying drawings. While the invention will be described inconjunction with these preferred embodiments, it will be understood thatit is not intended to limit the invention to such preferred embodiments.On the contrary, it is intended to cover alternatives, modifications,and equivalents as may be included within the spirit and scope of theinvention as defined by the appended claims. In the followingdescription, numerous specific details are set forth in order to providea thorough understanding of the present invention. The present inventionmay be practiced without some or all of these specific details. In otherinstances, well known mechanisms have not been described in detail inorder not to unnecessarily obscure the present invention.

It should be noted herein that throughout the various drawings likenumerals refer to like parts. The various drawings illustrated anddescribed herein are used to illustrate various features of theinvention. To the extent that a particular feature is illustrated in onedrawing and not another, except where otherwise indicated or where thestructure inherently prohibits incorporation of the feature, it is to beunderstood that those features may be adapted to be included in theembodiments represented in the other figures, as if they were fullyillustrated in those figures. Unless otherwise indicated, the drawingsare not necessarily to scale. Any dimensions provided on the drawingsare not intended to be limiting.

A method for dynamically enhancing audio is provided, for example, forselectively and dynamically expanding the dynamic range of the musicalpiece. In accordance with one embodiment, a static EQ is provided tostatically enhance a low and high band in conjunction with the dynamicenhancement.

The dynamic enhancement portion monitors the input energy level of afrequency band and responds to rapid changes in flux values. That is,rapid changes in the energy (e.g., note onsets, percussion hits,transients) results in modifications to the gain applied to low and highpass filters to boost the energy level for a limited period of time fromthe detection of the rapid change. An example non-limiting time rangefor the modification or enhancement is from 20-50 ms., but the scope ofthe invention covers both shorter and longer time periods.

The changes in flux values drive the characteristics of the time-varyingdynamic equalizer. Preferably, the analysis phase generates a controlsignal for control of the high and low pass filters, (e.g., dynamicshelving filters), positioned in series to modify the input audiosignals. For example, when there exists a transient in the lowfrequencies, the mapping of the control signal causes a boost for theshelving filter in the low band. A similar effect occurs when transientsor other energy level variations are detected within the high band.

FIG. 1 is a block diagram illustrating a system and method fordynamically enhancing an audio signal in accordance with one embodimentof the present invention. An input audio signal 102 is monitored todetect energy level variations. In a preferred embodiment, the energylevel variations are monitored using an analysis module 104 operating ona plurality of sampled values from the input audio signal. The analysismodule includes an analysis window that splits the incoming signal intooverlapping windows. Various types of analysis windows are known tothose of skill in the arts and hence complete details will not beprovided herein. Non limiting examples of suitable windows includerectangular windows and more preferably Hanning windows to avoidcreation of discontinuities in the combined signal.

The analysis window quantizes the input audio signal into a sequence offrames, each frame comprising at least one sample. In one embodiment,each frame comprises a plurality of samples with the samples groupedsequentially and presented for further analysis using an overlap and addprocedure. Overlap and add processing is known to those of skill in therelevant arts and hence complete details will not be provided herein. Inone embodiment, each frame comprises 256 samples with sequential framesoverlapping by 50%. In another embodiment, each frame comprises a singlesample.

Flux values derived from the analysis window 104 are mapped to a controlsignal 108 (i.e., a mapped flux value) to adjust the characteristics ofthe dynamic equalizer 106. Hence, the dynamic equalizer 106 is driven bythe signal itself. The analysis module 104 is configured to beresponsive to variations in signal energy, for example, sensitive tochanges in signal loudness. The control signal 108 provides continuouscontrol of the processing filters comprising the dynamic equalizer 106in order to provide an output signal selectively enhancing transients.

Further details regarding enhancement of an audio signal according tothe method described above are illustrated in FIG. 2. FIG. 2 is a blockdiagram illustrating a system for enhancing an audio signal inaccordance with one embodiment of the present invention. Preferably thesignal is split first into a low flux signal 206 using a low pass filter204 and into a high flux signal 212 using first a high pass filter 210.

According to an alternative embodiment, a frame of duration N of theinput signal 202 is pre-emphasized by applying a frequency band filter.Preferably separate frames for separate frequency bands are derived byapplying a first order high-pass filter 210 and a second order low passfilter 204. The frames are then fed respectively to modules that computeLPC coefficients using the autocorrelation method and theLevinson-Durbin recursion formula. The LPC coefficients comprise arepresentation of the audio signal 202.

Turning back to a preferred embodiment, two energy flux signals (206 and212) are first calculated from the original audio signal 202. These fluxsignals are designed to react respectively to low and high frequencytransients and are subsequently used as control signals in the dynamicgain and EQ stages. An energy flux represents the variation in time ofthe short-term energy of the incoming signal in low and high frequencyranges. The normalized flux f is then mapped via a non-linearsoft-decision function to create a control signal μ whose values vary,for a non-limiting example, from 0 to 0.5. An example transfer functionis illustrated in FIG. 4.

The first energy flux reacts to very low frequency transients (typicallykick drums), while the second one reacts to high-frequency transients(e.g., snare drums, and cymbals). Further details of an exampleembodiment for deriving the flux signals follow. Both flux signals areextracted from a “reference signal”: For a monophonic incoming signal,the reference signal is the signal itself, for a stereo signal it is thesum of the two channels, and for a multichannel signal, it is the sum ofthe left, right and center channels.

The low-frequency flux signal 206 is derived from an approximation ofthe RMS value of a low-pass version of the reference signal. Preferablythe low pass filter has a sharp cutoff. The lowpass filter 204 is in onevariation a third-order Chebyshev filter with a cutoff of 750 Hz. Thelow-frequency flux signal controls a dynamic second-order low-shelvingfilter 220 and a dynamic gain stage.

The high-frequency flux signal 212 is derived from an approximation ofthe RMS value of a high-pass version of the incoming reference signal.The highpass filter 210 according to this embodiment is a third-orderChebyshev filter with a cutoff of 12000 Hz. Both flux signals are thennormalized relative to the maximum of the reference signal over theanalysis frame. This renders them independent of the level of theincoming signal. The result of this normalization is further compressedin one embodiment by a sqrt( ) function, to compensate for the fluxsignals having too large a dynamic range.

Both flux signals are subjected to a non-linear filter whereby themaximum flux value from past frames (from indexes −40 to −2 in terms offrames, for example) is subtracted from the current flux value. This issomewhat akin to a first-order difference, but helps emphasize strongincreases even if they are not sudden (which is typically the case inthe low frequency flux).

In this preferred embodiment, there are additional non-linearitiesapplied to the flux signals. FIG. 4 is a diagram illustrating a transferfunction for mapping flux values to a control signal gain in accordancewith one embodiment of the present invention. Denoting f the flux signal(low or high frequency), the following non-linearities are applied:

f′=max(0,f−f0)

f″=min(f′*g,0.5)

where f0 represents a “floor” value, and g is a linear multiplicativefactor. The result f″ (the mapped flux value) is always between 0 and0.5. This means that the flux signals 402 are clamped to 0 when they'rebelow a floor value f0 (See reference number 403), a process which isintended to eliminate (usually noisy) values that are close to 0. Theflux signals are also smoothed by standard first-order non-linearfilters with a fast attack time and a slow release time. The slope 404of the transfer function 400 illustrated in FIG. 4 corresponds to asensitivity measure. Steeper slopes increase the sensitivity. Taken toan extreme value, a nearly vertical slope would tend to generate abinary treatment of the incoming flux. Hence, huge amplification wouldresult from a flux value slightly above the threshold, whereas noamplification would result from input flux values having a slightlysmaller magnitude. This would generate undesirable “pumping.” Oneskilled in the art could alter the slope of the transfer function inappreciation of the tradeoffs present and to avoid such undesirableartifacts. The mapped flux value 402 is capped at a maximum value 406,here predetermined to be a value of 0.5.

Turning back to FIG. 2, preferably, respective control signals μ 216(i.e., mapped flux values) are used to determine the gain for the lowshelving filter 220 and high shelving filter 222. A low frequency mappedflux 216 a (μ_(lp)) and a high frequency mapped flux 216 b (μ_(hp))control respectively the gains for the low shelving filter 220 and highshelving filter 222. In one embodiment, the shelving filters 220 and 222are 1^(st) order FIR shelving filters. In accordance with anotherembodiment, The low shelving filter 220 is a 2^(nd) order IIR filter andthe high shelving filter 222 is a first order IIR filter. In onenon-limiting embodiment, the low and high dynamic shelving filters (220,222) are both IIR filters based on the Regalia-Mitra topology. The scopeof the invention extends also to implementation with any other topology.

In this embodiment, the low-shelving filter 220 is a second order filterobtained by squaring a standard first-order Regalia-Mitra low-shelvingfilter having a gain denoted G_(lp). The second-order filter has a gaindenoted G_(lp) ² at DC, and 1 at Nyquist. The high-shelving filter 222is a first order Regalia-Mitra high-shelving filter with a gain denotedGhp at Nyquist. There is no limit imposed on the DC and Nyquist gains,so clipping can occur.

The dynamic parameters Glp and Ghp are directly controlled by the lowand high frequency flux signals using the following formula:

Glp=1+rμ _(ip)

Ghp=1+rμ _(hp)

where μ is the mapped flux signal (μ_(lp) is the low freq. mapped fluxand μ_(hp) is the high freq. mapped flux) and r is an adjustableparameter, i.e., a “sensitivity” control. As a result, the square of theDC gain of the low-shelving filter is roughly proportional to thelow-frequency flux signal, while the Nyquist gain of the high-shelvingfilter 222 is roughly proportional to the high-frequency flux signal.

The mapped flux signal (216 a, 216 b), i.e. a control signal, controlsthe gain that is applied by the second group of filters, i.e., filters220 and 222. In a preferred embodiment, presets are used to preselectthe applied slope and the thresholds, and hence affect the gain in thefilters 220 and 222. Preset options may be presented to a user through auser interface and may comprise any combinations of effects. Forexample, in one embodiment, 4 presets are presented to the user. The Lowpreset corresponds to a mild effect. The difference betweenbypass/non-bypass will be barely noticeable on most tracks, but thepresets might provide an impalpable general perceptual improvement ofthe audio. This would be the choice preset for someone who's keen onpreserving the authenticity of their audio tracks, but would beinterested in a slight overall improvement of the audio.

The Medium preset corresponds to a good balance between audibility ofthe effect, and naturalness. The effect will be noticeable yet notexcessive on most tracks. Percussions will sound sharper, high-hats,cymbal hits will be crispier, kick drum and snare drum hits will bepunchier, without sounding aggressive.

The High preset is intended to help demonstrate the effect of the audioenhancement processing. Exaggerating the modifications allows a user tobetter appreciate the audio enhancement capabilities. On most tracks,the effect will be very audible. Percussions will have a tendency tobecome aggressive on some tracks, kick drums almost abnormally loud andpunchy. On some mellow tracks, this preset will provide very pleasantresults, but on tracks that are already fairly punchy, the results willtend to be too aggressive.

The Game preset is recommended for game audio. This is the strongestpreset, and preserving the original quality of the background music isnot a primary goal. Rather, the emphasis is put on exaggerating audioeffects such as explosions, shots etc. If this preset is used on regularaudio tracks, the results will most likely sound unnatural. According toanother embedment, a slider control is provided in a user interface toenable the user to vary the extent of the audio enhancement. The slidercontrol preferably performs a linear interpolation on the gainparameters.

Blocks 224 and 226 provide synthesis and renormalization functionsrespectively. Their functions will be described in greater detail belowin the discussion regarding FIG. 3.

FIG. 3 is a flow chart illustrating a method of enhancing an audiosignal in accordance with one embodiment of the present invention.Initially, an audio input signal is received in operation 302. Next, afrequency band is identified in operation 304 for analysis. Preferably,the identification of the frequency band involves at least a lowfrequency band and a high freq. band. Although detection of high andlow-frequency transients and a subsequent engagement of a dynamic EQ toemphasize the corresponding frequency range are described, it should beunderstood that the invention is not limited to this example. The scopeof the invention covers analysis techniques applied to any and allcombinations of frequency ranges without limitation. Hence the scope ofthe invention includes but is not limited to analyses performed on 3 ormore frequency bands as well as an analysis performed on a single freq.band. In the latter case, the single frequency band can comprise theentire frequency spectrum of the incoming audio signal.

Next, the frequency limited band is sampled and windowed (e.g., ananalysis window applied) in operation 306. For example, a low passfiltered signal is segmented to generate a sequence of frames. Forexample, in one embodiment, the frame may comprise 256 samples.Preferably, the sampled values in the sequence of frames are groupedusing an overlap and add procedure, more preferably using an overlap of50%. It should be noted, however, that the scope of the invention is notso limited but rather extends to all variations of grouping the sampleddata and including without limitation all overlap and add techniques andpercentages of overlap, all variations of analysis windows applied toshape the frames, and all sizes of frames including analyses based on asample by sample basis.

In this embodiment, the method is operative to respond to the occurrenceof a transient (percussion hits, note onsets), to briefly engage adynamic EQ that emphasizes the corresponding frequency range. Theresult, for example, can include an increased crispness of the highfrequencies, more punchy mid-range percussions (snare drums, congas) andnote onsets, and stronger kick bass hits. The technique is based in thetime-domain rather than in the frequency-domain and hence has lowermemory and computational requirements than frequency domain basedanalyses.

Next, in operation 308, flux values are generated. That is, for eachanalysis frame derived, a flux value is generated. As used herein, fluxrefers to and represents the change in energy between the successiveframes in the plurality of frames derived in the preceding step. Inorder to determine the flux, an energy value is assigned to the frame orother sample grouping. Preferably the RMS value of signal is used todetermine the energy levels. The method examines the way the energylevel changes to generate a flux value for the frame. In one embodiment,applying something similar to a first order derivative, i.e., one frameless the previous frame, is performed to generate the flux value. Moredetails as to the frame comparison are provided below. In order todetermine the flux, only half (i.e., the most recent samples) of theframe are analyzed.

The flux correlates with transients. That is, the flux will yield a highenergy level for the frame when a transient occurs for that frequencyband under examination. For example, kick drums or cymbals will generatea transient and a corresponding peak in the flux.

Next, in operation 310 a mapped flux value is derived. A nonlinearfunction, such as illustrated in FIG. 4, will map that flux into amapped flux value for generation of coefficient for the correspondinglow shelving filter or high shelving filter. That is, for each separateframe, a mapped flux value is determined and provided as a controlsignal for controlling the coefficients and gain of the low and highshelf filters applied in series to the input audio signal.

In operation 312, the frequency limited filters are applied in series towindows of samples corresponding to the input audio signal. In apreferred embodiment, the frequency limited filters are low and highshelf filters. In another embodiment, the gain is applied to the entiresignal instead of a selected frequency band. Preferably, transientenhancement or modification occurs through the continuous recomputationof coefficients for digital filters applied to the incoming audiosignal. The coefficients for the filters are computed based on thesupplied control signal(s). Methods for determining filter coefficientsfrom gain values such as mapped flux values are known to those of skillin the relevant arts and hence complete details are not provided herein.

The filtering is a dynamic time varying process. In typical operation,the filter will remain constant over one frame and change for processingof the next frame. In a digital filter application, the coefficients arecomputed once per frame, applied to that frame of the input signal, andthen recomputed based on a new mapped flux value derived from the nextsignal frame. In a preferred embodiment, the gain is determined so as toboost the input audio signal upon detection of a transient.Alternatively, in accordance with another embodiment, the signal can bemade less percussive or punchy by attenuating the signal. In accordancewith yet another embodiment, a mechanism is provided such that the userselects whether gain or attenuation is the response to transientdetection.

Next, a synthesis window is applied in operation 314, followed by anoverlap and add procedure to reconstitute the modified or enhanced audiosignal. The synthesis window avoids discontinuities at the end of thewindow. Because the coefficients for both windows are not the same,i.e., the time varying filter will typically apply differentcoefficients to different frames, application of synthesis windows avoidundesirable artifacts (such as clicking). An overlap and add stepfollows to recombine the separate filtered windows or frames into theoutput audio signal.

Next, in operation 316, a normalization window is applied. This isdesirable to avoid clipping of the output signal and to ensure that theoutput signal precisely mimics the input signal in portions of the inputsignal where no enhancement is desired or not implemented. For theseunenhanced frames, the normalization window adjusts the output to matchthe input signal and is generally a function of the original shapes ofthe analysis windows used in operation 306.

In accordance with an alternative embodiment, the analysis and synthesiswindows (operations 306 and 314 respectively) are selected to result ina unity gain (i.e., gain=1) such that a renormalization window need notbe applied. The preferred embodiment, however, involves arenormalization window. This allows the user greater latitude in windowselection and compensates for the window shape through the use of therenormalization window.

Finally, in operation 320 an enhanced audio signal is provided as theoutput signal.

In an alternative embodiment, the time varying filter performs acontinuous filtering of the input signal by changing the coefficients ona sample by sample basis. Since the coefficients of the filter vary ateach sample, the signal may be reconstituted without using an overlapadd procedure operating on sequential frames. In one aspect, thetransient detection operates on a frame or block of samples and derivesa flux value for each sample by interpolating from the surroundingsample values (in the frame). The interpolated values are then used tomodify the filter coefficients.

That is, according to alternative embodiments, flux values for sample bysample filtering are derived by 1) computing the flux on a frame basisand interpolate; or 2) computing the flux on a sample basis, having onevalue per sample. In the latter alternative, the instantaneous energy iscomputed on a sample basis followed by smoothing and determination of afirst order difference.

The foregoing description describes several embodiments of a method forenhancing audio signals. A process is provided that is sensitive tovariations in signal loudness rather than to signal loudness itself.While the embodiments describe details of audio content sources, theinvention is not so limited but is intended to extend to all forms ofmedia signals such as including video signals. Further, while severalembodiments detailed herein describe enhancing the audio signal based oneither one or both of a low frequency band and a high frequency band,the scope of the invention is not so limited. The scope of the inventionincludes but is not limited to enhancing an audio signal in one band aswell as in 3 or more bands. In other embodiments, the audio enhancementoccurs in one or more low frequency bands, one or more mid bands, and/orone or more high frequency bands. Application of the audio enhancementtechniques are expected to be very effective in sharpening an audiotrack or brightening a sound, etc. In addition, a static equalizer maybeapplied to the audio regardless of the transients to provide anadditional brightening at both high and low frequencies.

Although the foregoing invention has been described in some detail forpurposes of clarity of understanding, it will be apparent that certainchanges and modifications may be practiced within the scope of theappended claims. Accordingly, the present embodiments are to beconsidered as illustrative and not restrictive, and the invention is notto be limited to the details given herein, but may be modified withinthe scope and equivalents of the appended claims.

1. A method for enhancing an audio signal comprising: monitoring thevariation in time of the energy level in at least one frequency band ofthe audio signal; and applying a dynamic time varying equalizer to theaudio signal based on the monitored energy level variations.
 2. Themethod as recited in claim 1 wherein monitoring the energy levelvariations comprises segmenting the audio signal into a sequence offrames, each frame having at least one sampled value; assigning anenergy level for each frame in the sequence; and deriving a flux valuecorresponding to the variation of the energy level between previousframes and the current frame.
 3. The method as recited in claim 2wherein the frames comprise only one sampled value and an energy levelis assigned to each sample.
 4. The method as recited in claim 2 whereinmonitoring the energy level variations comprises segmenting the audiosignal into a sequence of frames having at least one sampled value andassigning an energy level for each of a plurality of subsets of theframe.
 5. The method as recited in claim 4 wherein the subset of theframe contains only one sampled value and the energy level for eachsampled value is determined by interpolating the energy levels in atleast two adjacent frames in the sequence of frames.
 6. The method asrecited in claim 1 wherein the dynamic time varying equalizer comprisesat least one shelving filter having coefficient values determined by amapping process of the monitored energy level variations.
 7. The methodas recited in claim 1 wherein the at least one frequency band comprisesa low frequency band and a high frequency band.
 8. The method as recitedin claim 2 wherein the sequence of energy levels associated with theframes are used to generate flux values mapped to at least one controlsignal for controlling the dynamic time varying equalizer filter.
 9. Themethod as recited in claim 8 wherein the at least one control signalcomprises a mapped low frequency flux value and a mapped high frequencyflux value, the respective mapped flux values used to control a low freqshelving filter and a high freq. shelving filter in the dynamic timevarying equalizer filter.
 10. The method as recited in claim 8 furthercomprising combining a sequence of output values from the dynamic timevarying equalizer filter using an overlap and add procedure.
 11. Themethod as recited in claim 10 further comprising renormalizing thecombined sequence.
 12. The method as recited in claim 1 wherein theenergy level is determined based on the RMS value of the frame or framesubset examined.
 13. The method as recited in claim 1 wherein boostingthe level in the frequency band in the dynamic time varying equalizeroccurs when a flux value derived from the energy level variationsexceeds a predetermined threshold.
 14. The method as recited in claim 13wherein the transfer function between the flux and the control signal inthe filter in a nonlinear function.
 15. The method as recited in claim 2wherein the sequence of frames is generated by using a sampling window,boosting the level in the frequency band in the dynamic time varyingequalizer occurs when a flux value derived from the energy levelvariations exceeds a predetermined threshold, and the output signal fromthe dynamic time varying equalizer is generated by recombining frameswith a synthesis window, the configurations of the sampling window andthe synthesis window selected to generate a unity gain when the fluxfalls below a predetermined threshold.
 16. A system for enhancing anaudio signal comprising: a monitoring module configured to monitor thevariation in time of the energy levels in at least one frequency band ofthe audio signal; and a dynamic time varying equalizer controlled by themonitoring module.
 17. The system as recited in claim 16 wherein themonitoring module is configured to segment the audio signal into asequence of frames, each frame having at least one sampled value;assigning an energy level for each frame in the sequence; and deriving aflux value corresponding to the variation of the energy level betweenprevious frames and the current frame.
 18. The system as recited inclaim 17 wherein the dynamic time varying equalizer comprises a lowshelf filter and a high shelf filter, and wherein the coefficients forthe low shelf filter and the high shelf filter are derived from the fluxvalues.
 19. The system for enhancing an audio signal as recited in claim16 wherein the monitoring module is further configured to monitor a lowfrequency band of the audio signal and generate a first sequence of fluxvalues responsive to the variation in energy level of the audio signalover time for the low frequency band, monitor a high frequency band ofthe audio signal and generate a second sequence of flux valuesresponsive to the variation in energy level of the audio signal overtime for the high frequency band, generate a low frequency controlsignal and a high frequency control signal derived respectively from thefirst and second sequences of flux values; and further comprising: afiltering module having a low shelf filter and a high shelf filterpositioned in series to modify the audio signal, the gain for the lowshelf filter controlled by the low frequency control signal and the gainfor the high shelf filter controlled by the high frequency controlsignal.
 20. The system as recited in claim 19 wherein the filteringmodule is configured such that it applies a gain to a low frequency bandof the audio signal when the first sequence of flux values is responsiveto a low frequency transient and applies a gain to a high frequency bandof the audio signal when the second sequence of flux values isresponsive to a high frequency transient.
 21. A method for enhancing anaudio signal, the method comprising: segmenting the audio signal into asequence of frames; determining a flux value representing a timevariation in the energy levels between a first frame in the sequence andat least one of the preceding frames in the sequence; mapping the fluxvalue to a control signal for control of a filter configured to modifythe audio signal; and using the control signal to modify the gaincharacteristics of the filter, such that at least one frequency band ofthe audio signal is modified.
 22. The method as recited in claim 21wherein the mapping of the flux value to a control signal comprisesdetermining a sensitivity for the modifications based on a user input.23. The method as recited in claim 21 further comprising increasing theperceived level of loudness of the audio signal by performing at leastone of a bass and treble adjustment.